A. Field of the Invention
The present invention relates to a method and device for compensating for channel distortion in a communication receiver. Specifically the invention relates to an equalization method and structure for equalizing the receive data to compensate for channel distortion in DMT communication system as typically used in ADSL transceivers.
B. Description of the Related Art
1. Asymmetric Digital Subscriber Lines
Asymmetric Digital Subscriber Line (ADSL) is a communication system that operates over existing twisted-pair telephone lines between a central office and a residential or business location. It is generally a point-to-point connection between two dedicated devices, as opposed to multi-point, where numerous devices share the same physical medium.
ADSL supports bit transmission rates of up to approximately 6 Mbps in the downstream direction (to a subscriber device at the home), but only 640 Kbps in the upstream direction (to the service provider/central office). ADSL connections actually have three separate information channels: two data channels and a POTS channel. The first data channel is a high-speed downstream channel used to convey information to the subscriber. Its data rate is adaptable and ranges from 1.5 to 6.1 Mbps. The second data channel is a medium speed duplex channel providing bi-directional communication between the subscriber and the service provider/central office. Its rate is also adaptable and the rates range from 16 to 640 kbps. The third information channel is a POTS (Plain Old Telephone Service) channel. The POTS channel is typically not processed directly by the ADSL modemsxe2x80x94the POTS channel operates in the standard POTS frequency range and is processed by standard POTS devices after being split from the ADSL signal.
The American National Standards Institute (ANSI) Standard T1.413, the contents of which are incorporated herein by reference, specifies an ADSL standard that is widely followed in the telecommunications industry. The ADSL standard specifies a modulation technique known as Discrete Multi-Tone modulation.
2. Discrete Multi-Tone Modulation
Discrete Multi-Tone (DMT) uses a large number of subcarriers spaced close together. Each subcarrier is modulated using a type of Quadrature Amplitude Modulation (QAM). Alternative types of modulation include Multiple Phase Shift Keying (MPSK), including BPSK and QPSK, and Differential Phase Shift Keying (DPSK). The data bits are mapped to a series of symbols in the I-Q complex plane, and each symbol is used to modulate the amplitude and phase of one of the multiple tones, or carriers. The symbols are used to specify the magnitude and phase of a subcarrier, where each subcarrier frequency corresponds to the center frequency of the xe2x80x9cbinxe2x80x9d associated with a Discrete Fourier Transform (DFT). The modulated time-domain signal corresponding to all of the subcarriers can then be generated in parallel by the use of well-known DFT algorithm called Inverse Fast Fourier Transforms (IFFT).
The symbol period is relatively long compared to single carrier systems because the bandwidth available to each carrier is restricted. However, a large number of symbols is transmitted simultaneously, one on each subcarrier. The number of discrete signal points that may be distinguished on a single carrier is a function of the noise level. Thus, the signal set, or constellation, of each subcarrier is determined based on the noise level within the relevant subcarrier frequency band.
Because the symbol time is relatively long and follows a guard band, intersymbol interference is a less severe problem than with single carrier, high symbol rate systems. Furthermore, because each carrier has a narrow bandwidth, the channel impulse response is relatively flat across each subcarrier frequency band. The DMT standard for ADSL, ANSI T1.413, specifies 256 subcarriers, each with a 4 kHz bandwidth. Each sub-carrier can be independently modulated from zero to a maximum of 15 bits/sec/Hz. This allows up to 60 kbps per tone. DMT transmission allows modulation and coding techniques to be employed independently for each of the sub-channels.
The sub-channels overlap spectrally, but as a consequence of the orthogonality of the transform, if the distortion in the channel is mild relative to the bandwidth of a sub-channel, the data in each sub-channel can be demodulated with a small amount of interference from the other sub-channels. For high-speed wide-band applications, it is common to use a cyclic-prefix at the beginning, or a periodic extension at the end of each symbol, in order to maintain orthogonality.
In standard DMT modulation, each N-sample encoded symbol is prefixed with a cyclic extension to allow signal recovery using the cyclic convolution property of the discrete Fourier transform (DFT). Of course, the extension may be appended to the end of the signal as well. If the length of the cyclic prefix, L, is greater than or equal to the length of the impulse response, the linear convolution of the transmitted signal with the channel becomes equivalent to circular convolution (disregarding the prefix). The frequency indexed DFT output sub-symbols are merely scaled in magnitude and rotated in phase from their respective encoded values by the circular convolution. It has been shown that if the channel impulse response is shorter than the length of the periodic extension, sub-channel isolation is achieved. Thus, the original symbols can then be recovered by transforming the received time domain signal to the frequency domain using the DFT, and performing equalization using a bank of single tap frequency domain equalizer (FEQ) filters. The FEQ effectively deconvolves (circularly) the signal from the transmission channel response. This normalizes the DFT coefficients allowing uniform QAM decoding.
Such an FEQ is shown in FIG. 1. The FFT calculator 20 accepts received time domain signals from line 10, and converts them to frequency domain representations of the symbols. Each frequency bin (or output) of the FFT 20 corresponds to the magnitude and phase of the carrier at the corresponding frequency. In FIG. 2, each bin therefore contains a separate symbol value X(i) for the ith carrier. The frequency domain equalizer FEQ 40 then operates on each of the FFT outputs with a single-tap filter to generate the equalized symbol values Xxe2x80x2(i). The equalized symbol values may then be decoded by a slicer and demapper. The prior art frequency domain equalizer FEQ 40 of FIG. 2 suffers from technical disadvantages, as will become apparent in the following description.
An efficient and reliable method of using synchronization symbols to realize decision-feedback equalizer fault recovery while not disturbing the signal to noise ratio performance is provided. The method and apparatus uses aggressive decision-directed frequency domain equalizer (FEQ) adaptation during synchronization symbol intervals. It provides performance and reliability advantages in multicarrier communications systems such as ADSL modems.
The FEQ fault recovery is realized with aggressive forced adaptation using a derived signal during the ADSL synchronization symbol preferably only when a fault is detected. Faults are detected by calculating the minimum magnitude of constellation errors corresponding to faults when the showtime FEQ taps are used in the modified synchronization symbol receiver. The method utilizes the same FEQ taps that are used during normal data transmission mode (xe2x80x9cshowtimexe2x80x9d) during the synchronization symbol intervals. The taps are used to equalize the received synchronization symbols and the equalized output is compared to a derived reference synchronization signal. If a decision error would have resulted (e.g., in the showtime data slicer), it is determined that the the continued use of the FEQ taps would result in decision error propogation. The FEQ taps are therefore aggressively updated during the synchronization symbol interval. The FEQ tap updates can also be updated during a synchronization symbol using data re-use, i.e., performing multiple updates using the same synchronization symbol inputs. A data re-using LMS algorithm for the coefficient recursion is preferred, and it allows single synchronization symbol recovery when adequate computer run-time is available.
The fault recovery mechanism also allows more aggressive showtime tracking of any time variation of the channel because of the lessened concern of decision feedback error propagation. By calculating the magnitude of error signals corresponding to FEQ outputs, the forced adaptation of FEQ taps during the synchronization symbol can be disabled during fault-free operation, thereby reducing SNR performance degradation introduced by aggressive LMS updates. Alternatively, a normal update may be performed during the synchronization symbol.